Reduction of transient sounds in hearing implants

ABSTRACT

A method is described for generating electrode stimulation signals for electrode contacts in a cochlear implant electrode array. An input audio signal is processed to generate band pass channel signals that each represent an associated band of audio frequencies. A channel envelope is extracted from each channel signal. The input audio signal and the channel envelopes are processed to produce transient reduced envelopes based on: i. determining for each channel envelope a normalized channel-specific transient indicator characterizing transient noise present in the channel signal, ii. determining a combined transient indicator as a function of the channel-specific transient indicators, and iii. applying a channel-specific gain to the channel envelopes as a function of the combined transient indicator to produce the transient reduced envelopes. The transient reduced envelopes are then used to generate electrode stimulation signals to the electrode contacts.

This application is a continuation of co-pending U.S. patent applicationSer. No. 13/975,487, filed Aug. 26, 2013, and issued as U.S. Pat. No.8,929,994, which in turn claimed priority from U.S. Provisional PatentApplication 61/693,356, filed Aug. 27, 2012, which is incorporatedherein by reference.

FIELD OF THE INVENTION

The present invention relates to hearing implant systems such ascochlear implants, and specifically to the signal processing usedtherein.

BACKGROUND ART

A normal ear transmits sounds as shown in FIG. 1 through the outer ear101 to the tympanic membrane (eardrum) 102, which moves the bones of themiddle ear 103 (malleus, incus, and stapes) that vibrate the oval windowand round window openings of the cochlea 104. The cochlea 104 is a longnarrow duct wound spirally about its axis for approximately two and ahalf turns. It includes an upper channel known as the scala vestibuliand a lower channel known as the scala tympani, which are connected bythe cochlear duct. The cochlea 104 forms an upright spiraling cone witha center called the modiolar where the spiral ganglion cells of theacoustic nerve 113 reside. In response to received sounds transmitted bythe middle ear 103, the fluid-filled cochlea 104 functions as atransducer to generate electric pulses which are transmitted to thecochlear nerve 113, and ultimately to the brain.

Hearing is impaired when there are problems in the ability to transduceexternal sounds into meaningful action potentials along the neuralsubstrate of the cochlea 104. To improve impaired hearing, auditoryprostheses have been developed. For example, when the impairment isrelated to operation of the middle ear 103, a conventional hearing aidmay be used to provide acoustic-mechanical stimulation to the auditorysystem in the form of amplified sound. Or when the impairment isassociated with the cochlea 104, a cochlear implant with an implantedstimulation electrode can electrically stimulate auditory nerve tissuewith small currents delivered by multiple electrode contacts distributedalong the electrode.

FIG. 1 also shows some components of a typical cochlear implant systemwhich includes an external microphone that provides an audio signalinput to an external signal processor 111 where various signalprocessing schemes can be implemented. The processed signal is thenconverted into a digital data format, such as a sequence of data frames,for transmission into the implant 108. Besides receiving the processedaudio information, the implant 108 also performs additional signalprocessing such as error correction, pulse formation, etc., and producesa stimulation pattern (based on the extracted audio information) that issent through an electrode lead 109 to an implanted electrode array 110.Typically, this electrode array 110 includes multiple electrodes on itssurface that provide selective stimulation of the cochlea 104.

In cochlear implants today, a relatively small number of electrodes areeach associated with relatively broad frequency bands, with eachelectrode addressing a group of neurons through a stimulation pulse thecharge of which is derived from the instantaneous amplitude of theenvelope within that frequency band. In some coding strategies,stimulation pulses are applied at constant rate across all electrodes,whereas in other coding strategies, stimulation pulses are applied at anelectrode-specific rate.

Various signal processing schemes can be implemented to produce theelectrical stimulation signals. Signal processing approaches that arewell-known in the field of cochlear implants include continuousinterleaved sampling (CIS) digital signal processing, channel specificsampling sequences (CSSS) digital signal processing (as described inU.S. Pat. No. 6,348,070, incorporated herein by reference), spectralpeak (SPEAK) digital signal processing, and compressed analog (CA)signal processing. For example, in the CIS approach, signal processingfor the speech processor involves the following steps:

-   -   (1) splitting up of the audio frequency range into spectral        bands by means of a filter bank,    -   (2) envelope detection of each filter output signal,    -   (3) instantaneous nonlinear compression of the envelope signal        (map law).        According to the tonotopic organization of the cochlea, each        stimulation electrode in the scala tympani is associated with a        band pass filter of the external filter bank. For stimulation,        symmetrical biphasic current pulses are applied. The amplitudes        of the stimulation pulses are directly obtained from the        compressed envelope signals. These signals are sampled        sequentially, and the stimulation pulses are applied in a        strictly non-overlapping sequence. Thus, as a typical        CIS-feature, only one stimulation channel is active at one time        and the overall stimulation rate is comparatively high. For        example, assuming an overall stimulation rate of 18 kpps and a        12 channel filter bank, the stimulation rate per channel is 1.5        kpps. Such a stimulation rate per channel usually is sufficient        for adequate temporal representation of the envelope signal. The        maximum overall stimulation rate is limited by the minimum phase        duration per pulse. The phase duration cannot be chosen        arbitrarily short, because the shorter the pulses, the higher        the current amplitudes have to be to elicit action potentials in        neurons, and current amplitudes are limited for various        practical reasons. For an overall stimulation rate of 18 kpps,        the phase duration is 27 μs, which is near the lower limit. Each        output of the CIS band pass filters can roughly be regarded as a        sinusoid at the center frequency of the band pass filter which        is modulated by the envelope signal. This is due to the quality        factor (Q≈3) of the filters. In case of a voiced speech segment,        this envelope is approximately periodic, and the repetition rate        is equal to the pitch frequency.

In the existing CIS-strategy, only the envelope signals are used forfurther processing, i.e., they contain the entire stimulationinformation. For each channel, the envelope is represented as a sequenceof biphasic pulses at a constant repetition rate. A characteristicfeature of CIS is that this repetition rate (typically 1.5 kpps) isequal for all channels and there is no relation to the centerfrequencies of the individual channels. It is intended that therepetition rate is not a temporal cue for the patient, i.e., it shouldbe sufficiently high, so that the patient does not perceive tones with afrequency equal to the repetition rate. The repetition rate is usuallychosen at greater than twice the bandwidth of the envelope signals(Nyquist theorem).

Another cochlear implant stimulation strategy that transmits fine timestructure information is the Fine Structure Processing (FSP) strategy byMed-El. Zero crossings of the band pass filtered time signals aretracked, and at each negative to positive zero crossing a ChannelSpecific Sampling Sequence (CSSS) is started. Typically CSSS sequencesare only applied on the first one or two most apical channels, coveringthe frequency range up to 200 or 330 Hz. The FSP arrangement isdescribed further in Hochmair I, Nopp P, Jolly C, Schmidt M, Schöβer H,Garnham C, Anderson I, MED-EL Cochlear Implants: State of the Art and aGlimpse into the Future, Trends in Amplification, vol. 10, 201-219,2006, which is incorporated herein by reference.

FIG. 2 shows major functional blocks in the signal processingarrangement typical of existing cochlear implant (CI) systems whereinband pass signals containing stimulation timing and amplitudeinformation are assigned to stimulation electrodes. Preprocessor FilterBank 201 pre-processes an initial acoustic audio signal, e.g., automaticgain control, noise reduction, etc. Each band pass filter in thePreprocessor Filter Bank 201 is associated with a specific band of audiofrequencies so that the acoustic audio signal is filtered into some Nband pass signals, B₁ to B_(N) where each signal corresponds to the bandof frequencies for one of the band pass filters.

The band pass signals B₁ to B_(N) are input to a Stimulation PulseGenerator 202 which extracts signal specific stimulationinformation—e.g., envelope information, phase information, timing ofrequested stimulation events, etc.—into a set of N stimulation eventsignals S₁ to S_(N), which represent electrode specific requestedstimulation events. For example, channel specific sampling sequences(CSSS) may be used as described in U.S. Pat. No. 6,594,525, which isincorporated herein by reference.

Pulse Mapping Module 203 applies a non-linear mapping function(typically logarithmic) to the amplitude of the each band-pass envelope.This mapping function typically is adapted to the needs of theindividual CI user during fitting of the implant in order to achievenatural loudness growth. This may be in the specific form of functionsthat are applied to each requested stimulation event signal S₁ to S_(N)that reflect patient-specific perceptual characteristics to produce aset of electrode stimulation signals A₁ to A_(M) that provide an optimalelectric representation of the acoustic signal.

The Pulse Mapping Module 203 controls loudness mapping functions. Theamplitudes of the electrical pulses are derived from the envelopes ofthe assigned band pass filter outputs. A logarithmic function with aform-factor C typically may be applied to stimulation event signals S₁to S_(N) as a loudness mapping function, which generally is identicalacross all the band pass analysis channels. In different systems,different specific loudness mapping functions other than a logarithmicfunction may be used, though still just one identical function isapplied to all channels to produce the electrode stimulation signals A₁to A_(M) outputs from the Pulse Mapping Module 203.

Patient specific stimulation is achieved by individual amplitude mappingand pulse shape definition in Pulse Shaper 204 which develops the set ofelectrode stimulation signals A₁ to A_(M) into a set of output electrodepulses E₁ to E_(M) to the electrodes in the implanted electrode arraywhich stimulate the adjacent nerve tissue.

Background noise weakens the speech intelligibility of hearing aid andcochlear implant users. According to Hernandez et al., An Assessment OfEveryday Noises And Their Annoyance, Hearing Review, 2006, 13(7), 16-20(incorporated herein by reference), 33% of sensate background noise isformed by transient sounds such as computer key strokes, slamming doors,dish clattering, etc., all of which are unpleasant and reduce listeningcomfort (See also, German Patent DE 102005043314). The transient noisereduction algorithms in existing hearing aids such as the AntiShock fromUnitron Connect and the SoundSmoothing from Siemens have been found toyield an improvement in the listening experience. See DiGiovanni et al.,Effects of Transient-Noise Reduction Algorithms on SpeechIntelligibility and Ratings of Hearing Aid Users, American Journal ofAudiology, first published on Sep. 22, 2011 asdoi:10.1044/1059-0889(2011/10-0007), incorporated herein by reference.Transient noise reduction is also sought in other applications. Forexample, sound quality for car passengers may be improved by reducingthe transient road noise created when tires strike an obstruction. SeeU.S. Pat. No. 7,725,315, incorporated herein by reference. Likewise, inhigh-end audio equipment that renders audio data, the potential tomodify transient features like drumsticks hitting a drum is desired tomeet different individual preferences in music listening. See U.S. Pat.No. 7,353,169, incorporated herein by reference.

In existing cochlear implants, the incorporation of a dual front-endautomatic gain control (AGC) improves performance when intensetransients occur. See, e.g., Stöbich et al., Influence of Automatic GainControl Parameter Settings on Speech Understanding of Cochlear ImplantUsers Employing the Continuous Interleaved Sampling Strategy, Ear &Hearing, 1999, 20, 104-116, incorporated herein by reference. Howeverthe period of the AGC gain is too long to start a reduction at the onsetof the transients and the amount of reduction is not sufficient.

Transient signals are characterized by a fast and steep rising envelopeof the sound signal. Thus during the occurrence of a transient, theenvelope has much higher values for a short time interval. In GermanPatent DE 102005043314, the steepness and/or the amplitude of theenvelope of the sound signal are considered. If one or both of thesevalues exceed certain thresholds, the sound signal is attenuated.

In European Patent EP 1371263 (incorporated herein by reference), thesound signal is transformed into K sub-signals in the frequency domain.Then, for each sub-signal, two or three sub-indices are calculated whichare used to classify the present sound signal into the categories“stationary noise”, “quasi stationary noise”, “desired speech and music”and “transient noise”. These sub-indices refer to intensity changesduring a given time interval, the modulation frequency, and the durationof very similar intensities of the signal, respectively. According tothe classified category, a gain function is calculated, that is used tosuppress transient sounds or to enhance the SNR in case of theclassified categories “stationary noise” or “quasi stationary noise”.

In WO 99/53615 (incorporated herein by reference), a transient detectordivides the input signal into at least two frequency bands. In each ofthese bands, the derivative and/or the amplitude of the envelope arecompared to at least one threshold function to indicate a transient inthe respective band. If a transient is detected in at least one band,the coefficients of an adaptive filter are changed in such a way thatthe transients in the input signal are reduced by filtering the delayedinput signal with this determined adaptive filter. After the detector nolonger detects a transient, the filter coefficients return to the valuesbefore the transient has appeared.

In U.S. Pat. No. 7,353,169, the spectral flux is used to determinefrequency-specific indicators of transient features in high end audioequipment. According to these indicators, a modification of thecorresponding transient features is applied to improve the impression ofmusic. The user can decide on the amount, the frequency ranges, and thekind of modification (suppression or enhancement) he prefers.

U.S. Pat. No. 7,725,315 (incorporated herein by reference), describesusing models of transient road noise based on a code book or a neuralnetwork to attenuate transient sounds.

U.S. Pat. No. 7,869,994 (incorporated herein by reference) describes anattenuation of certain wavelet coefficients based on a threshold tosuppress transient sounds.

A possibility to reduce transient features in a cochlear implant systemis to use hearing aid algorithms as proposed in U.S. 2005/0209657(incorporated herein by reference).

In Stöbich 1999, a dual front-end AGC is proposed to reduce transientfeatures.

SUMMARY OF THE INVENTION

Embodiments of the present invention are directed to methods, systemsand software code for generating electrode stimulation signals forelectrode contacts in a cochlear implant electrode array. An input audiosignal is processed to generate band pass channel signals that eachrepresent an associated band of audio frequencies. A channel envelope isextracted from each channel signal. The input audio signal and thechannel envelopes are processed to produce transient reduced envelopesbased on: i. determining for each channel envelope a normalizedchannel-specific transient indicator characterizing transient noisepresent in the channel signal, ii. determining a combined transientindicator as a function of the channel-specific transient indicators,and iii. applying a channel-specific gain to the channel envelopes as afunction of the combined transient indicator to produce the transientreduced envelopes. The transient reduced envelopes are then used togenerate electrode stimulation signals to the electrode contacts.

The channel-specific transient indicator may be based on a proportion ofpower of the channel envelope to power of the input audio signal and/orhigh-pass filtering the channel envelope. The combined transientindicator may be based on a combined product of the channel-specifictransient indicators and/or a dependent function of the channel signals,which may reflect a limited frequency sub-range of the channel signals.

The channel-specific gains may be based on a single common gainfunction, a filter applied to the channel envelopes, and/or may reflecta signal-dependent suppression duration. A stationary noise reductionprocess may be applied to the channel envelopes before producing thetransient reduced envelopes.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 shows the anatomy of a typical human ear and components in acochlear implant system.

FIG. 2 shows major signal processing blocks of a typical cochlearimplant system.

FIG. 3 shows various functional blocks in a signal processingarrangement according to an embodiment of the present invention.

FIG. 4 is a graph showing an example of a speech input signal with twonoise transients.

FIG. 5 shows the effects of transient reduction in the frequency domain.

FIG. 6 shows examples of channel envelope signals in transientprocessing according to embodiments of the present invention.

FIG. 7 shows various functional blocks in a signal processingarrangement according to an embodiment including stationary noisereduction.

FIG. 8 shows application of dual front-end AGC to an audio signal.

DETAILED DESCRIPTION OF SPECIFIC EMBODIMENTS

Embodiments of the present invention are directed to reduction ofunpleasant transient sounds to improve the hearing comfort of cochleaimplant users and enhance speech intelligibility in environments withsignificant transient background noise such as a cafeteria. Simulationresults show that speech perception in quiet background conditions isunaffected.

FIG. 3 shows one specific embodiment for signal-processing in a cochlearimplant using a transient processing stage. A digitized input audiosignal s is processed by a filter bank 301 to generate K band passchannel signals that each represent an associated band of audiofrequencies. Instead of a time-domain filter bank 301 anotherpossibility to get a frequency domain sub-band splitting of the inputaudio signal s can be used, e.g., a FFT. Envelope modules 302 extract achannel envelope ENV1 to ENVK from each band pass channel signal.Transient reduction module 303 processes the input audio signal s andthe channel envelopes ENV1 to ENVK as will be discussed in detail belowto produce K transient reduced envelopes. Pulse generator 307 uses thetransient reduced envelopes to generate electrode stimulation signalsthat the transmitter 308 provides to the electrode contacts in theimplant 309.

FIG. 4 is a graph showing an example of a speech input signal with twonoise transients. The top plot shows an input speech signal with twonoise transients resulting from clattering which are located in the timeintervals [2.32, 2.37] and [2.63, 2.68] seconds. The second plot showsthe synthesized signal after passing the filter bank 301, and the thirdplot shows the resulting synthesized transient reduced envelope outputsignal from the transient reduction module 303. The bottom plot showsthe difference between these two synthesized signals showing that thespeech parts of the signals are not affected by the transient reductionmodule 303, only the two noise transients are reduced.

FIG. 5 demonstrates the effect of the transient reduction in thefrequency domain. The top image shows the spectrogram of the synthesizedsignal after the filter bank 301, which is presented in the second plotof FIG. 4, where it is clear that the main energy of the noisetransients is located in the high frequency regions. The second image inFIG. 5 shows the spectrogram of the synthesized transient reducedenvelope output signal from the transient reduction module 303, and thebottom image shows the spectrogram of the quotient of these twosynthesized signals in which it is clear that the speech features arepreserved while the high frequency elements of the noise transients arereduced.

Considering the transient reduction module 303 in greater detail,normalized indicator modules 304 receive the input audio signals s andthe corresponding k-th channel envelope to produce normalizedchannel-specific transient indicators characterizing transient noisepresent in the channel signal. These can be determined as:

$v_{k} = {a_{k} \cdot \left( \frac{{envelope}_{k}}{z} \right)^{2}}$where a_(k) is a non-negative channel-specific parameter which controlsthe size of v_(k) depending, for instance, on the settings of the filterbank 301. The signal z results from low-pass filtering and rectifyingthe signal s, i.e., z=LP(|s|). The normalization of the envelope withthe signal z is necessary because then v_(k) describes the proportion ofthe power of a transient signal in the k-th channel envelope related tothe power of the whole signal. Moreover, the normalization ensures thatthe reduction of the noise transient is independent of the loudness ofthe audio input signal s.

The top row of FIG. 6 shows the input envelopes of channels 9 to 12resulting from the audio input signal s that is plotted in the top ofFIG. 4. The boundary frequencies of these channels are 2294, 3201, 4445,6153 and 8500 Hz. The second row in FIG. 6 presents the correspondingchannel-specific transient indicators v₉ to v₁₂. Only the locations ofthe noise transients v_(k) all have large values. Instead of using theenvelope value to determine the channel-specific transient indicators v₁. . . , v_(K), a specific embodiment could use a high-pass filteredenvelope, for example, the first derivative of the envelope. In additionor alternatively, the information of both features can be used todetermine the channel specific indicators v₁ . . . , v_(K), that is, acombination of the value and the high-pass filtered value of theenvelope.

Combined transient indicator module 305 receives as inputs thechannel-specific transient indicators v₁, . . . , v_(K) and develops anoutput signal combined transient indicator w. Noise transient signals(e.g., from dish clattering or rustling paper) typically have highenvelopes in all signal channels higher than approximately 1 kHz. Thus,the channel-specific transient indicators v_(k) of these channels alsohave high values. This is not the case for consonants and plosives suchas “s”, “sch”, “t”, “tz” where only some of the channel-specificindicators have high values. Given a set of channels: M={j: the lowerboundary frequency of channel j is greater than 1 kHz}, then a highvalue of the signal

$w = {\prod\limits_{j = M}\; v_{j}}$relates to the presence of a noise transient signal, whereas thecombined transient indicator w has relatively low values in the cases ofconsonants, plosives and fricatives. The third plot in FIG. 6 shows theindicator

${w = {\prod\limits_{j = 9}^{12}\; v_{j}}},$which is greater than 0 at the positions of the onset of the noisetransient.

Instead of the multiplication

${w = {\prod\limits_{j = M}\; v_{j}}},$an embodiment could use any function ƒ(v₁ . . . , v_(K)) with thefollowing properties:

-   -   Only the selected set of channels M={j: channel j is located        within a certain frequency range} influence the result.    -   If one of the channel-specific transient indicators v_(k), kεM        has low values, then ƒ is small, too.    -   If all the channel-specific transient indicators v_(k), kεM have        high values, then ƒ is high.    -   If all the channel-specific transient indicators v_(k), kεM/{j}        have constant values greater than zero, then ƒ is a monotone        increasing function of v₁.        The selected set of channels M can differ between the output        channels. This means for example, distinguishing between        transients in the low, middle and high frequency channels to        reduce the corresponding low, middle and high transient        features. Then the combined transient indicator module 305 has        multiple combined transient indicator outputs w_(k).

Channel-specific gain module 306 receives the combined transientindicator w and the corresponding envelope of the k-th channel toproduce transient reduced envelope signals. Channel specific gain aredetermined and applied to the channel envelopes to suppress noisetransients. Depending on the combined transient indicator w, an actualgain value is determined: g=max(1−σ·w,l), where 0<l≦1 is the lower boundof the suppression factor g and σ is a channel-specific positiveconstant parameter which determines the amount of the suppression in thechannel. Next, the gain function h is calculated. This function shouldimmediately reduce the noise transients when they occur, but the gainfunction h also should increase with an exponential decay (fast attack,slow release). This leads to the following approach:h[n]=(1−b _(r))·h[n−1]+b _(r) ·g[n], if h[n−1]<g[n] (release)h[n]=b _(a) ·h[n−1]+(1−b _(a))·g[n], if h[n−1]≧g[n] (attack)with 0≦b_(a),b_(r)<<1. Note that a time-index n is included since afeedback loop exists. A small value of b_(a) results in a fast decay ofh[n]. Thus, the reduction of the transient signal starts immediately. Ifh[n−1]<g[n], then the suppression factor h increases slowly asdetermined by the release-time constant b_(r). The transient reducedoutput envelopes are then generated by multiplying h by the inputenvelope signals. The bottom row of FIG. 6 shows the resultingtransient-reduced envelopes.

Instead of the calculation of one gain function h, coefficients of alinear FIR filter or a nonlinear filter can be calculated that areapplied to the envelope signal. The method for the calculation of thegain can be modified in such a way that the duration of the suppressionis signal dependent, e.g., replacing the parameter b_(r) by a functionof b_(r)(w, v₁, . . . , v_(K)). The attack time then depends on theconstant parameter b_(a). This could be changed by modifying thecalculation of the gain function or by a signal dependent parameterb_(a). The application of the gain to the envelope can be different froma simple multiplication, for example a FIR filter or an N-of-M typecochlear implant coding strategy can be controlled by the combinedtransient indicator w.

FIG. 7 shows a noise reduction arrangement which includes a stationarynoise reduction module 701 in front of the transient reduction module303. The signals from the stationary noise reduction module 701influence the determination of the combined transient indicator w. Soif, for example, a voice activity detector indicates the presence ofspeech, then it is assumed that a speech feature is present in thatchannel and this is not reduced by the transient reduction.

The foregoing transient noise reduction techniques are different fromthe other earlier arrangements discussed in the Background sectionabove. In DE 102005043314, the reduction of transients is done in thetime domain without considering frequency specific features; i.e., theprocessing is done without splitting the signal into frequency parts.Furthermore, a threshold is used to determine if the signal has atransient feature, which is not the case in the above described method.

In EP 1371263 a classification is performed into the categories“stationary noise”, “quasi stationary noise”, “desired speech and music”and “transient noise”. And furthermore, the sub-indices to classify thesignals are different what is described above.

WO 99/53615 uses a threshold to indicate a transient signal. And only asingle gain is applied to the input signal s, whereas the embodimentsdiscussed above apply the channel-specific gains on each of the channelenvelopes.

In U.S. Pat. No. 7,353,169 the spectral flux constitutes a norm over thefrequencies at each time of the first derivate in time. These normsdiffer from what is described above that uses multiplication over thefrequencies, which is not a norm.

U.S. Pat. No. 7,725,315 uses special features of transient noise in acar to detect transients via a codebook or a neural network. U.S. Pat.No. 7,869,994 uses a wavelet transformation. These are completelydifferent compared from the transient reduction described above. In US2005/0209657 no algorithm is proposed to reduce noise transient signals,and the only discussion is of using in cochlear implants the algorithmsemployed by hearing aids.

Stöbich 1999 proposed using a dual front-end AGC to reduce transientfeatures. FIG. 8 shows the result of the dual front-end AGC. The topplot shows the input signal which is a mixed audio signal of speech withthree instances of a dish clattering (shaded grey). In the middle andbottom plots the output signal and the corresponding gain of the AGCrespectively are shown. It can be seen that the gain is reduced when anoise transient occurs, but the onset is missed and the amount of thereduction is not sufficiently high compared to the suppression resultsin the example in FIG. 5.

The prior art does not describe normalization with the low-pass filteredsignal z, and most of the other approaches use a threshold to decide ifa noise transient is contained in the audio input signal. And theembodiments of the present invention described above refrain completelyfrom using any kind of threshold.

In certain cases, an embodiment may erroneously detect consonants asnoise transients, undesirably damping such consonants and impairingtheir perception. Simulation results yielded a maximum false detectionrate below 5 percent if a stationary noise reduction algorithm is addedinto the signal processing in front of the transient reduction module.For bilaterally implanted users, the interaural level differences can bechanged in certain cases, degrading the localization of transient sounds

Embodiments of the invention may be implemented in part in anyconventional computer programming language. For example, preferredembodiments may be implemented in a procedural programming language(e.g., “C”) or an object oriented programming language (e.g., “C++”,Python). Alternative embodiments of the invention may be implemented aspre-programmed hardware elements, other related components, or as acombination of hardware and software components.

Embodiments can be implemented in part as a computer program product foruse with a computer system. Such implementation may include a series ofcomputer instructions fixed either on a tangible medium, such as acomputer readable medium (e.g., a diskette, CD-ROM, ROM, or fixed disk)or transmittable to a computer system, via a modem or other interfacedevice, such as a communications adapter connected to a network over amedium. The medium may be either a tangible medium (e.g., optical oranalog communications lines) or a medium implemented with wirelesstechniques (e.g., microwave, infrared or other transmission techniques).The series of computer instructions embodies all or part of thefunctionality previously described herein with respect to the system.Those skilled in the art should appreciate that such computerinstructions can be written in a number of programming languages for usewith many computer architectures or operating systems. Furthermore, suchinstructions may be stored in any memory device, such as semiconductor,magnetic, optical or other memory devices, and may be transmitted usingany communications technology, such as optical, infrared, microwave, orother transmission technologies. It is expected that such a computerprogram product may be distributed as a removable medium withaccompanying printed or electronic documentation (e.g., shrink wrappedsoftware), preloaded with a computer system (e.g., on system ROM orfixed disk), or distributed from a server or electronic bulletin boardover the network (e.g., the Internet or World Wide Web). Of course, someembodiments of the invention may be implemented as a combination of bothsoftware (e.g., a computer program product) and hardware. Still otherembodiments of the invention are implemented as entirely hardware, orentirely software (e.g., a computer program product).

Although various exemplary embodiments of the invention have beendisclosed, it should be apparent to those skilled in the art thatvarious changes and modifications can be made which will achieve some ofthe advantages of the invention without departing from the true scope ofthe invention.

What is claimed is:
 1. A computer program product implemented in anon-transitory, tangible computer readable storage medium for generatingelectrode stimulation signals for electrode contacts in a cochlearimplant electrode array, the product comprising: program code forprocessing an input audio signal to generate a plurality of band passchannel signals each representing an associated band of audiofrequencies; program code for extracting a channel envelope from eachchannel signal; program code for processing the input audio signal andthe channel envelopes to produce transient reduced envelopes based on:i. determining for each channel envelope a normalized channel-specifictransient indicator characterizing transient noise present in thechannel signal, ii. determining a combined transient indicator as afunction of the channel-specific transient indicators, and iii. applyinga channel-specific gain to the channel envelopes as a function of thecombined transient indicator to produce the transient reduced envelopes;and program code for using the transient reduced envelopes to generateelectrode stimulation signals to the electrode contacts.
 2. A productaccording to claim 1, wherein the channel-specific transient indicatoris based on a proportion of power of the channel envelope to power ofthe input audio signal.
 3. A product according to claim 1, wherein thechannel-specific transient indicator is based on high-pass filtering thechannel envelope.
 4. A product according to claim 1, wherein thecombined transient indicator is based on a combined product of thechannel-specific transient indicators.
 5. A product according to claim1, wherein the combined transient indicator is based on a dependentfunction of the channel signals.
 6. A product according to claim 5,wherein the function reflects a limited frequency sub-range of thechannel signals.
 7. A product according to claim 1, wherein thechannel-specific gains are based on a single common gain function.
 8. Aproduct according to claim 1, wherein the channel-specific gains arebased on a filter applied to the channel envelopes.
 9. A productaccording to claim 1, wherein the channel specific gains reflect asignal-dependent suppression duration.
 10. A product according to claim1, further comprising: program code for applying a stationary noisereduction process to the channel envelopes before producing thetransient reduced envelopes.
 11. A signal processing arrangement forgenerating electrode stimulation signals for electrode contacts in acochlear implant electrode array, the arrangement comprising: a filterbank pre-processor configured to process an input audio signal togenerate a plurality of band pass channel signals each representing anassociated band of audio frequencies; a channel envelope moduleconfigured to extract a channel envelope from each channel signal; atransient reduction module configured to process the input audio signaland the channel envelopes to produce transient reduced envelopes basedon: i. determining for each channel envelope a normalizedchannel-specific transient indicator characterizing transient noisepresent in the channel signal, ii. determining a combined transientindicator as a function of the channel-specific transient indicators,and iii. applying a channel-specific gain to the channel envelopes as afunction of the combined transient indicator to produce the transientreduced envelopes; and a stimulation signal generator configured to usethe transient reduced envelopes to generate electrode stimulationsignals to the electrode contacts.
 12. An arrangement according to claim11, wherein the transient reduction module determines thechannel-specific transient indicators based on a proportion of power ofthe channel envelope to power of the input audio signal.
 13. Anarrangement according to claim 11, wherein the transient reductionmodule determines the channel-specific transient indicator based onhigh-pass filtering the channel envelope.
 14. An arrangement accordingto claim 11, wherein the transient reduction module determines thecombined transient indicator based on a combined product of thechannel-specific transient indicators.
 15. An arrangement according toclaim 11, wherein the transient reduction module determines the combinedtransient indicator based on a dependent function of the channelsignals.
 16. An arrangement according to claim 15, wherein the functionreflects a limited frequency sub-range of the channel signals.
 17. Anarrangement according to claim 11, wherein the transient reductionmodule bases the channel-specific gains on a single common gainfunction.
 18. An arrangement according to claim 11, wherein thetransient reduction module bases the channel-specific gains on a filterapplied to the channel envelopes.
 19. An arrangement according to claim11, wherein the transient reduction module applies channel specificgains that reflect a signal-dependent suppression duration.
 20. Anarrangement according to claim 11, further comprising: a stationarynoise reduction module before the transient noise reduction moduleconfigured to apply a stationary noise reduction process to the channelenvelopes.